Clean up abs-send-time for audio.

webrtc/voice_engine/include/voe_rtp_rtcp.h

Index: webrtc/voice_engine/include/voe_rtp_rtcp.h
diff --git a/webrtc/voice_engine/include/voe_rtp_rtcp.h b/webrtc/voice_engine/include/voe_rtp_rtcp.h
index e344d0488c62053a2238daa694033177d88df19a..716566de32dc3fa9e199d77de3881caf9d83c7c3 100644
--- a/webrtc/voice_engine/include/voe_rtp_rtcp.h
+++ b/webrtc/voice_engine/include/voe_rtp_rtcp.h
@@ -133,16 +133,6 @@ class WEBRTC_DLLEXPORT VoERTP_RTCP {
     return 0;
   }
 
-  // Sets the status of sending absolute sender time on a specific |channel|.
-  virtual int SetSendAbsoluteSenderTimeStatus(int channel,

[the sun, 2016/10/27 16:09:17]

Do you need to PSA this change to the legacy, but public, API?

[stefan-webrtc, 2016/10/31 15:40:27]

In theory, yes, but I'm pretty sure it's being unused. I could send a PSA, but I
don't think it's worth waiting 2 weeks...

-                                              bool enable,
-                                              unsigned char id) = 0;
-
-  // Sets status of receiving absolute sender time on a specific |channel|.
-  virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
-                                                 bool enable,
-                                                 unsigned char id) = 0;
-
   // Sets the RTCP status on a specific |channel|.
   virtual int SetRTCPStatus(int channel, bool enable) = 0;